Methods and apparatus for forwarding IP calls through a proxy interface

ABSTRACT

A session controller configured to be coupled to a communications network is disclosed. The session controller is configured to receive an indication of a Voice over Internet Protocol (VoIP) registration request at a first time from an endpoint having a third-party registration, and to receive a signaling message from the endpoint at a second time later than the first time. The session controller is further configured to forward an indication of the VoIP registration request and a signaling message to an application server, such that at least a portion of the topology of the communications network beyond the session controller is hidden to the endpoint.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. patent application Ser. No.11/026,746, filed Dec. 31, 2004, and International Application No.PCT/US2005/018870, filed May 27, 2005, which claims priority to U.S.patent application Ser. No. 11/026,746, filed Dec. 31, 2004, bothentitled, VOICE OVER IP (VOIP) NETWORK INFRASTRUCTURE COMPONENTS ANDMETHOD, the entire content of which is hereby incorporated by reference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates generally to a method and apparatus forcarrying real time services, such as voice telecommunication, via apacket switched network and in particular to an apparatus and method forvoice, facsimile and multimedia over Internet Protocol (IP)communications components.

2. Description of the Related Art

Voice telecommunications has traditionally been conducted via dedicatedtelephone networks utilizing telephone switching offices and eitherwired or wireless connections for transmitting the voice signal betweenthe users' telephones. Such telecommunications, which use the PublicSwitched Telephone Network (PSTN), may be referred to as circuitcommitted communications. Voice over Internet Protocol (VoIP) providesan alternative voice telecommunication means which use discrete packetsdigitized voice information to transmit the voice signals. The packetsare transmitted either over the public Internet or within intranets.

Typical VoIP network infrastructure includes gateways, gatekeepers,proxy servers, softswitches, session border controllers, etc. Due tooptimization of network resources and to particular designs, networkoperators may choose to integrate functionality of the separatecomponents with one another such that multiple infrastructure componentscan be collocated on one physical component.

It is desirable that the VoIP network infrastructure components bedesigned into a network such that network operators can providemeaningful services to their customers.

The following terms are used in this disclosure:

Gateway—An entity that can bridge or serve as a “gateway” betweennetworks. In VoIP, it typically refers to a device that can “gateway”between the traditional Public Switched Telephone Network (PSTN) and theVoIP network.

Gatekeeper—An entity that works in conjunction with the gateway todetermine how to handle VoIP calls. The gatekeeper can be either in thecall path or play only a consultative role in every call. The gatekeeperusually only handles VoIP calls setup using the H.323 protocol.

Proxy Server—An intermediate entity, similar in functionality to thegatekeeper, that determines how to handle VoIP calls. A proxy serverusually only handles VoIP calls setup using the session initiationprotocol (SIP).

User Agent—An entity that can place or receive a VoIP call, usuallybased on the SIP protocol (session initiation protocol).

Border element—A border element is also called network edge element.This is typically where the policy definitions or the administrativecontrol changes. Policy can be defined at virtually all layers in theseven layer open systems interconnection (OSI) model. For example, atlayer three of the seven layer model policy can typically be describedin terms of routing peers, advertised IP routes etc. Routers wouldtypically act as the border elements where such policies change betweennetworks. Network address translators (NATs) act as border elements toconnect two or more non-routable address domains. Firewalls implementpolicy control (for layer three and above) as border elements where theadministrative control changes. The application layer typically usesflows at lower layers as well (for example, in the network layer and thetransport layer). Control of the application layer potentially allowscontrol of microflows at lower layers. For example, individual mediastreams for SIP calls having identical layer three characteristics maybe subject to different policies. Session layer border control (SBC)allows other border elements (like routers, NAT/Firewalls, and qualityof service brokers) to understand these microflows and provide theappropriate policy on a more granular basis. As a stand-alone element,an SBC simply allows policy control at the application layer.

Subnet—A subnet is an IP (Internet protocol) subnetwork inside a realm

Call Peer A call peer is a logical grouping for calls. Call peers may bestatic (created by the administrator) or dynamic (created at runtime bythe multi-protocol session controller). A call peer must belong to asingle device and may belong to one or more call peer groups. There aretwo kinds of call peers: an ingress call peer and an egress call peer,as defined in the following.

Ingress Call Peer—An ingress call peer is a call peer which isassociated with the incoming of a call.

Egress Call Peer—An egress call peer is a call peer which is associatedoutgoing of a call.

Call Peer Group—A call peer group is a (logical) grouping of call peersbased on policy (business policy, for example, service level assurancesor allocation of enterprise resources), for example, sites or peers.

Device—A device is a collection of call peers. A device may be static(have a fixed binding between call peers and a layer three address) ordynamic (when protocol registrations are to create the binding betweencall peers and layer three addresses). A dynamic device may have staticor dynamic call peers. A static device only has static call peers.

Template—A template is a rule set used for dynamically managing devicesand call peers, such as subnets.

IWF—SIP/H.323 Inter-working Function

A-O-R SIP—Address of Record (RFC 3261)

AAA—Authentication, Authorization and Accounting. These refer to thethree functions performed for every call to authenticate a user's phonecall, authorize the user to utilize resources in the network and accountfor the resource usage.

SUMMARY OF THE INVENTION

The present invention provides infrastructure such that networkoperators can enable their services to be delivered to other networkoperators, to other enterprise customers, as well as to residentialcustomers. This includes carrier-carrier peering, carrier-enterprisepeering, and carrier-residential peering, respectively.

The present invention provides a system that includes sessioncontrollers for packet switched voice telecommunications, including amulti-protocol signaling switch and a multi-protocol session controller,and a comprehensive management system for the session controllers. Themanagement system is able to provision information into the sessioncontrollers, as well as to report on the operation of the sessioncontrollers.

A family of session controller (SC) products is preferably providedalong with a comprehensive management system for the sessioncontrollers. The management system is able to provision information intothe session controller, as well as to report on the operation of thesession controller.

When incorporated into an overall architecture of the network thesession controller typically processes calls and hence, participates inall calls that flow through it. Every call processed by the sessioncontroller produces a call detailed record (CDR) that is stored locallyon the session controller until it is securely and reliably transportedto operations support systems (OSS) and/or to the management system. Themanagement system also receives a copy of every call detailed recordproduced. An analytics engine (AE) of the management systems processesthe call detailed records to produce engineering reports, generatealarms and exceptions, and to produce routing rules for the sessioncontroller based on business policy.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a schematic representation of a telecommunications systemarchitecture according to the principles of the present invention;

FIG. 2 is a block diagram of the elements of a multi-protocol sessioncontroller 25 * policy database, including call peers, groups, devicesand templates;

FIG. 3 is a block diagram showing the association of one ingress callpeer and one egress call peer in the call routing process;

FIG. 4 is a block diagram of a database policy model for the callrouting process;

FIG. 5 is a block diagram of call routing on a multiprotocal sessioncontroller;

FIG. 6 is a block diagram of an error code handling process;

FIG. 7 is a block diagram of a system for simultaneously deployingsession and border control;

FIG. 8 is a block diagram of a session control architecture;

FIG. 9 is a block diagram of policy;

FIG. 10 is a schematic diagram of a call admission control on a peergroup basis;

FIG. 11 is a schematic diagram of a media routing policy configuration;

FIG. 12 is a schematic diagram of a firewall traversal on amulti-protocol session controller;

FIG. 13 is a schematic diagram of a separation between signally andmedia on a far-end network address translator traversal;

FIG. 14 is a schematic diagram of a first scenario for a network addresstranslator traversal trough a single firewall;

FIG. 15 is a schematic diagram of is a second scenario for a networkaddress translator traversal trough a distributed firewall;

FIG. 16 is a-schematic diagram of is a third scenario for a networkaddress translator traversal through two firewalls;

FIG. 17 is a schematic diagram of a multi-protocol session controller asan SDX gateway client;

FIG. 18 is a schematic representation of shared transcoding resources ata network core;

FIG. 19 is a diagram of the physical architecture of a multi-protocolsession controller according to the present invention;

FIG. 20 is a schematic view of a session controller within a trustednetwork providing media communications to non-trusted network;

FIG. 21 is a schematic representation of a measure of call quality usingthe present session controller; and

FIG. 22 is a schematic representation of an in memory database.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 illustrates an overall network architecture of a system usingsession controllers and a management system of the present invention toprovide voice over Internet protocol telephone service. The term voiceover Internet protocol (VoIP) includes not only voice communications butalso includes fax data, multimedia data and other real time or near realtime services. These services may also be known collectively as mediacalls. The system, denoted generally at 20 in the figure, provides aninterface between a public switched telephone network (PSTN) 22, abroadband network address translator (NAT) traversal 24, enterprisepeering 26, an H.323 network 28, and a session initiation protocol (SIP)softswitch network 30. By way of explanation, the public switchedtelephone network 22 is the traditional public telephone system usingcircuit switched voice networks. An H.323 network 28 refers to a networkutilizing H.323 standards for packet-based transfer of information,including voice transmissions, and in particular to the interfacebetween the circuit switched voice transmissions and the packet switchedvoice transmissions. For example, the H.323 network may be the Internet.A softswitch network refers to a network using switches in which thereis a separation of the network hardware from the network software. TheSIP softswitch network and H.323 network utilize carrier peering.Peering refers to the exchange of information via nodes in a networkwithout a central controller using the same protocol layer as otherunits in the communication system. Enterprise peering refers to suchcommunications within an enterprise.

The public switched telephone network 22 interfaces through a softswitch32 to the system 20. The interface between the broadband network addresstranslator (NAT) traversal 24 and the system 20 is via broadband system34 and a multi-protocol session controller (MSC) 36. The enterprisepeering system 26 communications through the present system 20 via amulti-protocol session controller (MSC) 38. A single multi-protocolsession controller (MSC) 40 is provided for the H.323 network 28 and thesession initiation protocol (SIP) softswitch network 30.

Within the system 20, the multi-protocol session controllers 36, 38 and40 and the softswitch 32 communicate with a multi-protocol signalingswitch (MSW) 42. The multi-protocol signaling switch 42 and themulti-protocol session controllers 36, 38 and 40 are controlled by amanagement system 44. The management system 44 is operable to provisioninformation into the session controllers 36, 38 and 40, as well as toreport on the operation of the session controllers. The sessioncontrollers 36, 38 and 40 process calls and participate in all callsthat flow through the respective session controller. The calls processedby the session controllers 36, 38 and 40 are documented in a call detailrecord (CDR) that is stored locally on the session controller and istransmitted to the management system. The management system alsoreceives a copy of every call detail record produced by each of thesession controllers and processes the call detail records to produceengineering reports, generate alarms and exceptions, and produce routingrules for the session controller based on business policy.

The session controllers process calls that are either at the edge of thenetwork, or at the core of the VoIP network. If a session controller isplaced at the edge of the VoIP network, peering with other networks,then the function is called border session controller (BSC) or sessionborder controller (SBC). A session controller at the core of the networkprovides functions such as call routing and aggregate call admissioncontrol (CAC) and is referred to as a core session controller (CSC).

The multi-protocol session controllers 36, 38 and 40 can act either asthe core session controller or as the border session controller in anetwork. The following briefly describes the functions that aresupported.

In a deployment of a session controller as a core session controller,for example, the controller performs call routing and serves as ahunting engine. When deployed as a core session controller, it alsofunctions as a core call controller, wherein all calls are routedthrough the multi-protocol session controller. The multi-protocolsession controller 36, 38 and 40 identifies calls which need externalfeature and application servers. Further, the multi-protocol sessioncontroller 36, 38 and 40 can act as the session initiation protocol(SIP) outbound proxy for endpoints accessing services on an applicationserver. Another function performed is authentication, authorization andaccounting (AAA), wherein the multi-protocol session controller 36, 38and 40 can enable authentication, authorization and accounting using aremote authentication dial-in user server (RADIUS). Local passwordauthorization and call detail record logging can also be used. The coresession controller also performs core network call admission control,such as regulating network capacity. The multi-protocol sessioncontroller 36, 38 and 40 exchanges telephony routing informationprotocol (TRIP) messages with other domains to advertise and learnroutes. Lastly, the multi-protocol session controller can act as the3GPP S-CSCF (Third Generation Partnership Project Service Call SessionControl Function).

In a second deployment scenario, the session controller functions as aborder session controller where it performs topology hiding. Themulti-protocol session controller 36, 38 or 40 can be engaged ininter-working function calls, typically between SIP (Session InitiationProtocol) and H.323 networks. This is because the multi-protocol sessioncontroller 36, 38 or 40 can connect to the various access networks whichhave H.323 entities in them. The conversion between SIP and H.323 isreferred to the inter-working function (IWF). The inter-working functionis well known for voice calls. The present session controller includesan inter-working function for video calls. In particular, the presentsession controller has support for video calls between H.323 endpointsand SIP endpoints, as well as having such inter-working function supportfor voice calls.

The multi-protocol session controller 36, 38 or 40 will also beproviding interoperability functions in the network. The controllerperforms access network call admission control, such as bandwidthcontrol. For networks directly registering to it, the multi-protocolsession controller can enable call screening and user authentication.The multi-protocol session controller 36, 38 or 40 can use RADIUS(Remote Authentication Dial-In User Server) or DIAMETER (a protocolsimilar to RADIUS, for authentication, authorization and accounting)based SIP (Session Initiation Protocol) authentication.

An additional function of the multi-protocol session controller 36, 38or 40 is to control media flows between the access networks. Themulti-protocol session controller provides far-end network addresstranslation traversal for the access network. The multi-protocol sessioncontroller can provide transcoding services in the network for callsgoing out to the access network. The transcoding media server resourcesmay be centralized in the network or co-located with the multi-protocolsession controller (for example, at the border). The multi-protocolsession controller also acts as the interception related information(IRI) intercept access point and content intercept access point forCALEA (the Communications Assistance for Law Enforcement Act (of 1994)).Additionally, the multi-protocol session controller 36, 38 or 40provides monitoring for call jitter and for the mean opinion score forthe media streams.

Quality of service monitoring is performed by the border sessioncontroller, wherein the multi-protocol session controller acts as aDiffserv (differentiated services) border element. The differentiatedservices code point (DSCP) is controllable on a per call basis. Themulti-protocol session controller also takes care of inserting and/ormodifying the virtual local area network tags and priority bits. Thevirtual local area network tags are controlled on a per realm basis andthe virtual local area network priority values are controlled on a percall basis. The controller also provides the 3GPP (Third GenerationPartnership Project) proxy call session control function (P-CSCF), theinterrogating call session control function (I-CSCF) and the breakoutgateway control function (BGCF). Lastly, as the border controller, themulti-protocol session controller can act as the SIP outbound proxy forendpoints registered to a third party application server.

All the multi-protocol session controllers (both the border controllersas well as the core controllers) in the network download theirrespective databases from the centralized partitioned database schemastored on the iVMS management system (a proprietary management system ofthe assignee of the present application). For accounting purposes, themulti-protocol session controllers can also be integrated with a RADIUSserver (Remote Authentication Dial-In User Server) for accounting. AnSNMP (Simple Network Management Protocol) agent runs on themulti-protocol session controller as well as on the iVMS managementsystem.

FIG. 2 shows the multi-protocol session controller policy database.Templates 50 are used to manage dynamic devices 52 (which aredynamically created devices), call peers 54 and bindings 56. A dynamicassociation with call peer groups 58 using the templates 50 is indictedby the line 60. Static devices 62 are indicated

Call peers 54 are by far the most important element in the processing ofa call. As mentioned above, a call peer is a logical grouping of calls,and can be either an ingress call peer of incoming calls or an egresscall peer of outgoing calls. The multi-protocol session controllerassociates exactly one ingress call peer for each call being processed.The process of call routing ensures that exactly one egress call peercan be associated with the call.

FIG. 3 shows this conceptually. The incoming or ingress call peer 68 ismatched to a call peer 70 provisioned in the database. The ingress callpeer 68 is instantiated by the multi-protocol session controller bymostly looking at the protocol message which triggered the call/sessionprocessing. Information at layers three through seven of the OSI sevenlayer model is analyzed to create this instantiation. The outgoing oregress call peer 72 is instantiated by the routing policy 74 configuredin the database. In particular, once the ingress call peer 68 iscreated, the multi-protocol session controller looks for a match forthis call peer in the database (or policy). If a match is not found, thecall may still be accepted if access control has been disabled. Undernormal configuration, however, such calls are dropped. The process ofcall routing loops through the database and instantiates the egress callpeer 72 for the call leg. The multi-protocol session controller may notbe able to associate an egress call peer 72 for a call leg if resourcesare not available or user policy prevents it. Under such circumstancesthe call is rejected.

As an intermediate system that does not originate or terminate phonecalls, the present system sits in the path of the phone call andswitches the call. Each call has two call legs, an incoming call leg andan outgoing call leg when seen from the intermediate components.

The process of matching an ingress call peer 68 with one provisioned inthe policy 74 depends on how call peers, devices and templates aredefined. The matching process results in the allocation of the ingresscall peer 68 with a call peer defined in the database and hence allowsthe latter to define policy on groups of calls.

In the following, it is assumed that the term call peer refers to thecall peer defined in the database. The process of call routing is simplyrepresented in FIG. 4.

Some examples of call peers 68 are: calls identified by the called partynumber (or pattern) or the calling party number (or pattern); callsoriginating from a single signaling entity, for example an H.323-ID oran address-of-record; or calls already grouped by trunks groups.

A device contains one or more call peers (and contains at least one).The containment relationship is referred to as binding. Both the deviceas well as this binding can be static or dynamic. Devices usuallycorrespond to physical things in the network around the multi-protocolsession controller, such as gateways, terminals, gatekeepers, multi-portconferencing units (MCUs), conferencing servers, private branchexchanges (PBXes), proxy servers, etc.

The association of a call to a device is not a hard and fast rule. Thefollowing restrictions do apply. Each registration must correspond to aunique device. The multi-protocol session controllers will not allow adevice to register successively or in multiple registration messages.This simplifies registration management, caching, and timeouts ofdevices which use registration.

A registration activates all call peers on a device which usesregistration. Until the first registration arrives, the call peers,device and the bindings are considered disabled.

Examples of device-based identification criteria used by themulti-protocol session controllers are: the resolved value of the viaaddress in the first session initiation protocol (SIP) INVITE(identifying the previous hop proxy—hop to hop identification); theresolved value of the contact uniform resource identifier (URI) in thefirst SIP INVITE or source signaling address in the H.323 SETUP(identifying the previous hop back to back user agent or the end useragent/endpoint/gateway—end to end identification); and the source IP(Internet protocol) address of the incoming SIP INVITE or H.323 SETUP(identifying the previous hop network address translation or middlebox).

A call peer group 58 allows multiple call peers to be part of a commonpolicy specification.

The templates 50 have multiple functions. In particular, templates 50function as rule sets which govern the application of policy to devicesand to call peers which are not explicitly provisioned on themulti-protocol session controller. Templates 50 function as rule setswhich govern the application of policy to devices which register withthe multi-protocol session controller. Both of these functions areexplained in detail hereinafter.

For every call, the existence of the call peer at ingress allows thefollowing functions to be performed.

Call admission control is based on call legs or bandwidth. If accesscontrol is enabled and no ingress call peer is found or no templateexists which allows the creation of a call peer dynamically, the call isrejected. If a call peer is found, call legs and bandwidth used arecompared against limits.

The call peer at ingress also permits the peer to be reported in thecall detail record for every call. Further, a media routing policy isprovided for the call. In addition, a privacy policy for the call (SIPPrivacy and 11.323 presentation and screening indicators) are provided.

The call peer allows interworking of related information for originatingor terminating signaling on the peer. Specifically, the calling peerspecifies what kind of signaling (SIP or H.323) is used to terminatecalls on the peer (this applies to device peers, see below).

Yet another function permitted by the call peer is the specification ofthe class of service (in a tiered service model) for calls originatingfrom the call peer. The class of service is specified using trunkgroups, zones, call hunting timeouts and attempts, etc.

There are also limitations in some embodiment, such as themulti-protocol session controller lacks extensive support for devicepeers and subnet peers in the current releases. However most of theseconcepts are visible in the current database and policy settings. Areg-id/u-port acts as a call peer and the reg-id (registryidentification) functions as the device holder. There is also no clearseparation between devices and call peers. Configuration specific todevices is part of each call peer which is part of the device. Forexample, the IP address information is statically configured on allreg-id/u-ports which are assumed part of the same device. An i-edgegroup is the only call peer group defined.

Further, the multi-protocol session controller does not require explicitprovisioning of devices and call peers. In the scenario where SIP useragents access services on a third party SIP server using themulti-protocol session controller as the outbound proxy, themulti-protocol session controller does not require the user agents to beexplicitly provisioned as call peers or devices. These devices mustregister with the server and the multi-protocol session controllercreates dynamic devices and call peers based on these registrations. Themulti-protocol session controller also instantiates policy for thesedynamic devices/peers using templates. In this case, templates 50 (basedon subnets) may be used to dictate creation of these dynamic call peersand devices and their association with i-edge groups. Each template hasa reg-id/u-port.

Gateways/user agents may sometimes be moved (providing mobility). Themulti-protocol session controller detects mobility only viaregistrations. Mobile devices may be registered (and hence configured)on the multi-protocol session controller or a third party server.Templates 50 govern the instantiation of policy on both cases.

In the following is described the policy configuration on a call peer.

The multi-protocol session controller policy and configurationparameters on each call peer are assumed to be in four broad categories.The categorization is two ways, namely IN, representing parameters whichapply for incoming calls or those defined on an ingress call peer, andOUT, representing parameters which apply for outgoing calls or thosedefined on egress call peers. Within each of these categories there aretwo categories, namely MATCH, parameters which are used for matchingcalls (These may be protocol parameters, layer three (network)parameters or application aliases.) or SET, parameters which are usedfor modification of call parameters (These may be protocol parameters,layer three parameters or application aliases.).

The use of these parameters is explained further in the call routingsection. The IN/MATCH parameters function to associate a call with aningress call peer. When the same parameter is associated with multiplecategories, for example IN/MATCH as well as IN/SET, it is assumed to bea unique instance of the parameter in each category which is unrelatedto other categories.

Dynamic call peers are call peers that are created dynamically by themulti-protocol session controller when acting as the SIP outbound proxy.This is explained later in this specification.

The following describes use cases involving templates.

In an example including mobility, network address translation (NAT) andoutbound proxy (OBP), a mobile endpoint registers with an address insubnet1. The endpoint is always behind a network address translator(NAT) (along with several other similar endpoints) and is registeringwith a third party SIP server through the multi-protocol sessioncontroller. The endpoint later moves to subnet2. In this case, themulti-protocol session controller creates a device and call peerdynamically for each registration and applies a subnet specific policyto instantiate the call peer. Each such endpoint behind the networkaddress translator has a unique reg-id (registry identification) that isdynamically generated by the multi-protocol session controller.

In an example that is the same as above with the exception of theoutbound proxy, the endpoint is registering with the multi-protocolsession controller. In this case, the user has a predefined reg-id(registry identification) for the endpoint. Templates based on subnetsgovern policy instantiation when the endpoint moves.

In another example, a gateway within a subnet attempts to make a call.Subnet based templates allow the multi-protocol session controller toassociate a call peer as well as a call peer group with the call.

The call processing algorithm, or method, used by the multi-protocolsession controller as shown in FIG. 5 includes four steps. In Step 1,labeled IF, reference number 80, filter the call on the ingress (usingthe IN/MATCH parameters above). In Step 2, labeled IX, reference number82, translate the call on the ingress (using the IN/SET parametersabove). In Step 3, labeled EF, reference number 84, filters the call onthe egress (using the OUT/MATCH parameters above). In Step 4, labeledEX, reference number 86, translates the call on the egress (using theOUT/SET parameters above).

In further detail, step 1, 80, is the process of source identificationand applying common source policy. Steps 2 and 3, 82 and 84, togetheraccomplish “routing” of the call. Step 2, 82, is also called tagging andstep 3, 84, is called matching. Thus, routing is the combined process oftagging and matching. Step 4, 86, accomplishes hand-off of the call.

All steps except step 3 (routing) are optional. Generally, there are twokinds of calls. The first type of call include calls which are directedto a specific endpoint (for example, to an address of record, aparticular phone or to a particular H.323 identification). These callsexecute step three only and are referred to as direct end point calls.The second type of calls include calls which require call hunting on themulti-protocol session controller. Call hunting is the process by whichthe multi-protocol session controller finds the best possibledestination among an identified group of destinations for the call.

Call hunting uses the following criteria: filter priority; call-peerpriority; filter match strength; administration policies on call-peer orfilter (like time of day filtering); load balancing (least recently usedor percent utilization); and run time criteria like ISDN/SIP responsecodes/redirects, etc. Similarly, source identification uses thefollowing: filter priority; call-peer priority; filter match strength;and administration policies on call-peer or a filter (like time of day).

The control of error codes returned by the multi-protocol sessioncontroller is as follows. Both SIP and H.323 calls use several errorcodes to signal why a call is dropped. The multi-protocol sessioncontroller allows control of the call hunting process based on the errorcodes and the mapping of the error codes when they are returned back tothe caller. Generally, an error code can be interpreted: as a stop(wherein no more attempts are necessary for the call and themulti-protocol session controller also uses this interpretation fordirect end point calls which fail); as a temporary failure (keeptrying); or as a redirect (try the attached list of destinations).

FIG. 6 illustrates the error code handling process.

The call hunting policy 88 is applied before the error code 90 is mapped92 and sent to the caller. No error code mapping 92 is done for directend point calls. The multi-protocol session controller uses a defaultpolicy on how the error code is used by the call hunting policy. Bydefault, no error code mapping is applied unless protocol inter-workingis necessary. The inter-working error code map is also defined as partof the full error code map on the in or ingress peer (device peer).

The following describe call routing and hunting. The policy describedabove enables the following modes of routing:

Automatic number identification (ANI) based routing: The multi-protocolsession controller is capable of doing automatic number identificationbased routing in a multitude of ways. If the ANI policy is specific to acall origination, but common to all terminations, then the IN/MATCHparameters are used to filter the automatic number identification at theingress (Step 1 of call routing above). The call is tagged (Step 2).This tag is used as the filter for the OUT/MATCH step which follows. Thetag applied in the call tagging step is filtered (this is the step 3 ofcall routing). Translations may be applied as part of Step 1 and Step 4in call routing.

If the automatic number identification policy is specific to a calltermination, but common to all originations, then the OUT/MATCHparameters are used to filter the automatic number identification at theegress (Step 3 of call routing). Translation may be applied as part ofstep 4 in call routing.

The multi-protocol session controller can use the automatic numberidentification to identify the true origination of the call. Forexample, in the scenario where the multi-protocol session controllerfunctions as a session initiation protocol outbound proxy, a calloriginated by an endpoint registered to a session initiation protocolserver may get hair-pinned through the multi-protocol sessioncontroller. The hair-pinned instance may have no session initiationprotocol headers in common with the original call coming in (for examplewhere the session initiation protocol server is a back to back useragent). For such a call, the multi-protocol session controller can usethe automatic number identification as a selector identifying the trueaccess network originating the call. In this way, if the call isdestined towards any public switched telephone network gatewayregistered to the multi-protocol session controller, the policy can beselected based on the originating access network. Automatic numberidentification based call routing can also used by the multi-protocolsession controller on calls coming in from the access networksthemselves to determine if they need any application services. Callswhich do not need application services may be directly routed to otherregistered endpoints (such as the public switched telephone networkendpoints).

In trunk group based routing, the trunk groups can be used on themulti-protocol session controller in a multitude of ways. First, anorigination can specify a termination policy by providing themulti-protocol session controller with a trunk group identifier. Second,for class of service, the tagging step in call routing process can beused to tag calls based on a required class of service. Thirdly,partitioned routing or interconnects is used somewhat like a dumb patchpanel, where a call origination is connected with one or moreterminations. Fourthly, for simple automatic number identification ordialed number identification service based routing, using the filteringmechanism (step 1 of routing), calls can be tagged based on automaticnumber identification or dialed number identification service. Step 3then chooses a termination based on these tags.

All call-peer ports can be placed into zones. A caller which is in zoneA is allowed to speak only to other parties in the same zone.

Control of the call hunting process is provided in the multi-protocolsession controller. In particular, control specific to call origination(characterizing policy applied to a call source) is provided. Thisincludes the maximum number of attempts allowed for a call source and amaximum post-dial delay specified for the source. The post-dial delay(PDD) timer may also expire when a destination responds with a callproceeding or a 100 trying. 100 trying is a SIP message code formattedaccording to RFC822. In this case, the multi-protocol session controllerwill abandon the call attempt.

Further, the multi-protocol session controller controls the call huntingprocess by providing that any Internet control message protocol (ICMP)destination that is unreachable may be coming in response to a pendingrequest (both SIP and H.323).

Another mechanism for control of the call hunting process provides for ahunt timeout specified on a destination basis (SIP and H.323). Thistimeout determines when the multi-protocol session controller considersan attempt failed and tries the next alternate route.

As a further control of call hunting, sticky routes are used. A stickyroute is defined as the last route used in the call hunting process. Inessence it is used to terminate the call in case the attempt which usesthe sticky route fails. Sticky routes function as good exceptionmechanisms to a general hunt process and allow a call termination todetermine and terminate the call hunting algorithm.

Filters and Translations which are common to all calls can be applied aspart of transit routes (special kind of routes).

The multi-protocol session controller interacts with an externalapplication server.

For endpoints that are registered directly with an application server,the multi-protocol session controller directly hands off calls to theapplication server. The application server may direct the call to avoice mail server (which may pass through the multi-protocol sessioncontroller) or to another registered endpoint through the multi-protocolsession controller (hairpin).

The multi-protocol session controller also creates a dynamic endpointstate for each registration destined to the application server. Thedynamic state is created on the realm which the registration comes onand all calls originating from and terminating to these endpoints assumethe media routing characteristics of the realm.

The multi-protocol session controller also uses the session initiationprotocol mirror proxy functionality to achieve the same. Themulti-protocol session controller will allow assignment of a mirrorproxy on a call-peer basis.

The multi-protocol session controller routes registrations and calls forall endpoints using the mirror proxy functionality to their respectivemirror proxy (provisioned at the source of call identified by themulti-protocol session controller).

The multi-protocol session controller also functions as a sessioninitiation protocol (SIP) outbound proxy. The concept of the outboundproxy is defined for the SIP protocol only and applies to calls orservices being accessed off a third party SIP server using themulti-protocol session controller as an intermediate element.

An ingress session initiation protocol call/registration is classifiedby the multi-protocol session controller into two categories. First, themulti-protocol session controller serves as the proxy/registrar. In thiscase, the call/registration is accessing authentication/routing servicesof the multi-protocol session controller. A request for a uniformresource identifier for registrations must be addressed to themulti-protocol session controller. However, request for a uniformresource identifier for calls may not be addressed to the multi-protocolsession controller. The multi-protocol session controller also processesSIP 3xx messages locally without passing them on to the caller.

Second, the multi-protocol session controller serves as the outboundproxy. For a registration, the request uniform resource identifier isaddressed to a third party SIP server. All calls coming from suchendpoints and not addressed to the multi-protocol session controller aretreated as outbound proxy calls. Calls addressed to the multi-protocolsession controller are still routed as in the proxy mode above. Themulti-protocol session controller always relays all SIP final responses(including the 3xx message codes, which relate to redirection responsesof the SIP messages) back to the caller in this case. As a consequenceof this, no hunting services are provided on the multi-protocol sessioncontroller as well. (This is to make sure authentication works properlyon each hunt attempt as well as that there is no undue effect on thehunting algorithm implemented on the external server).

As described above, the multi-protocol session controller has theability to act as a SIP outbound proxy in both the border as well ascore of the session control. A border controller acting as the SIPoutbound proxy would forward registrations and signaling messagesdirectly to the end server. Location of such an end server (using adomain name server (DNS), for example RFC 3263) is typically hidden fromthe endpoints using the multi-protocol session controller as theoutbound proxy. The multi-protocol session controller can also employcall hunting to hunt through a locally configured list of servers aspart of the location process. Processing of SIP 3xx redirect messagecodes is also executed on the multi-protocol session controller.

The multi-protocol session controller can treat a call as an outboundproxy call in two ways. The first way uses SIP request—uniform resourceidentifier based forwarding for calls from endpoints which areregistered with a third party registrar. In this method, the end systemaccessing the SIP server is aware of the existence of the multi-protocolsession controller as the outbound proxy. This method is advisable whenthe multi-protocol session controller executes the core session controlfunction (as a core controller). The second way uses mirror proxyfunctionality on the multi-protocol session controller. In this method,the end system accessing the SIP server presumes that service isprovided by the multi-protocol session controller. Note that the mirrorproxy functionality only applies to endpoints which are registered. Themulti-protocol session controller forwards all registrations andsignaling messages transparently between the end system and the server.This method is preferable when the multi-protocol session controllerexecutes the border session control function (as a border controller).The mirror proxy functionality allows more control by the administratorover non-conforming SIP endpoints.

FIG. 7 illustrates a scenario where border control and core sessioncontrol are not on the same element, both methods may be simultaneouslydeployed in the system. In particular, the border controller 100 deploysa proxy to the core controller 102 that in turn provides the proxy tothe application server 104.

In FIG. 8, the session control architecture, also referred to as aprotocol stack, provides that the border session control function (BSCF)Policy 106 provides the border session control function (BSCF) 108.Likewise, the core session control function (CSCF) Policy 110 providesthe core session control function (CSCF) 112. The BSCF 108 effects theH.323 routed gatekeeper (GK) 114, the session initiation protocol(SIP)/H.323 inter-working function (IWF) 116, the session initiationprotocol (SIP) proxy/registrar 118, and the session initiation protocol(SIP) outbound proxy (OBP) 120. The core session control function (CSCF)112, on the other hand, effects only the SIP proxy/registrar 118 and theSIP OBP 120.

A mirror proxy may be deployed on the border controller. Arequest—uniform resource identifier based forwarding is deployed in thecore. Call hunting is also executed at the core.

The following discloses the creation and management of dynamic peers.

The multi-protocol session controller uses dynamic peers (also known asdynamic endpoints) to manage registrations and calls for user agentswhich register with a third party registrar. Both the device and thecall peer are created and managed at runtime. The following stepsdescribe how this state is managed.

In the first step, the state is created on a successful registrationwhen the registrar returns a 200 OK. The 200 OK refers to the SIPmessage code indicating that the response has been successfullyprocessed. A temporary state is maintained while the registration is inprogress. The profile used to create the dynamic call peer and devicemay vary depending on whether a template is discovered for it or whetherdefaults are being used.

In the second step, the state is refreshed on every registration. Themulti-protocol session controller maintains a timeout based on the 200OK message sent back for a register. If the endpoint does not refresh,it is deleted.

In the third step, the state is deleted on an unregistration.

In the fourth step, the multi-protocol session controller always assignsa timeout value (in response to a 200 OK message) which is the minimumof the locally configured value and that assigned by the registrar.

In the fifth step, the network address translation state is stored inthe dynamic device and maintains signaling information for routing callsback to the user agent.

In the sixth step, an implicit access control may be enabled for alldynamic call peers by limiting the previous hop for routing calls to theuser agent. The multi-protocol session controller allows the previoushop to be open or restricted to the proxy/registrar to which theendpoint registers.

In the seventh step, the uniform resource identifier with which the useragent can be accessed on the multi-protocol session controller is of theform: user@MSC-Realm, where the registration is for user@proxy.

In the eighth step, each dynamic call peer and device corresponds to aunique session initiation protocol registration and has a unique reg-id(and a port of zero). This allows the user agent to be mobile from onenetwork to another, especially when there is a network addresstranslation between the user agent and the multi-protocol sessioncontroller.

In the ninth step, a dynamic endpoint belongs to the realm on which theregistration arrives. When the endpoint moves and the realm changes, themulti-protocol session controller will update it on the next successfulregistration.

The templates provide the following functionality. A dynamic call peeris created when the third party session initiation protocol registrarresponds with a 200 OK message for the registration. An administratormay associate policy which applies to these dynamically created peers.For example, depending on which subnet the registration originates from,it is within the scope of the invention to associate calls coming fromthese devices to a site specific media or call admission control policy.When a template is not found, the multi-protocol session controllerpreferably creates the device and call peer using default parameters.

Templates have two main functions. The first is assignment of policy toinactive devices and the bound call peers based on registrationinformation. The second is assignment of policy to non-existent devicesand call peers based on registration information.

For example, a template can contain a subnet IP address and mask as itsIN/MATCH criteria. On the first registration from this subnet, there aretwo possibilities:

The call peer is located and holds the registration alias. In this case,the multi-protocol session controller may use the templates IN/SET andOUT/SET parameters to modify the existing parameters on the device andcall peers. This would be used, for example, in case a user agent movesfrom one subnet to another and the system applies a new media routingpolicy to calls coming from it.

When a call peer is non-existent as well as the device then themulti-protocol session controller would use the IN/SET and OUT/SETparameters to instantiate the call peer and device. The IN/MATCH andOUT/MATCH parameters would be initialized by the multi-protocol sessioncontroller based on the protocol parameters and state created. Thiswould be used, for example, when session initiation protocol user agentsare registering to a third party session initiation protocol server. TheOUT/MATCH parameter in this case would be the session initiationprotocol A-O-R=user@MSC-Realm. The IN/MATCH parameter would be thesession initiation protocol contact address=user@Private-Address and isused to group calls coming from the user agent.

The present multi-protocol session controller (MSC) also addressesbilling issues. The multi-protocol session controller can function asthe central point in the network which routes all calls. Call detailedrecords are produced for each call and logged using ASCII/RADIUS. Forcalls (identified by Call ID or Callid) which are hairpinned by anapplication server (AS), the call flow appears as follows:

Callid1→MSC Callid2→AS→Callid3-+MSC→Callid4

The call detailed records (CDRs) produced for such a call are asfollows:

CDR1 (on MSC): Callid1, Callid2

CDR2 (on AS): Callid2, Callid3

CDR3 (on MSC): Callid3, Callid4.

Extra CDR desired: Callid1, Callid4. All of these call detailed recordsare desirable to be able to debug and account for all call legs.

To mediate the call detailed records to be able to produce a single calldetailed record, several approaches can be taken. The multi-protocolsession controller can implement the IMS charging ID (3GPP) and insertit as part of the P-Charging-Vector (RFC 3455). The application servermust support this header and relay it back to the multi-protocol sessioncontroller. This is the best solution. Unfortunately, it requires thatthe application server (which will be a back to back user agent in mostcases) pass this header on, unmodified. In this case, the multi-protocolsession controller will specially mark CDR1-3 as the intermediate calldetailed record (CDR) and produce the final call detailed record.

The three call detailed records can be mediated on an external system toproduce the final call detailed record.

A topology hiding function is provided for all endpoints directlyregistered to the multi-protocol session controller. The multi-protocolsession controller provides these functions for calls going betweenrealms, calls within a realm, media flowing between realms, and mediaflowing within a realm (if necessary).

Under the heading of inter-working function and interoperability, thefollowing apply. For border control, the multi-protocol sessioncontroller uses the session initiation protocol back to back user agentand the H.323 back to back gateway as the architectural components.

The H.323 routed gatekeeper 114 accesses an H.323 gatekeeper 122 and anH.323 gateway 124. The SIP/H.323 inter-working function accesses theH.323 gateway 124 and an SIP user agent (UA) 126. The SIP user agent isalso referred to using session description protocol (SDP) or SIP-T(session initiation protocol for telephony). The SIP proxy/registrar 118accesses a back to back user agent 128 that sits atop the SIP user agent126, as well as accessing an SIP registrar 130. The SIP outbound proxy(OBP) 120 only accesses the back to back user agent 128.

The H.323 gatekeeper 122, H.323 gateway 124, SIP user agent 126 and SIPregistrar 130 form a layer that sits atop a layer formed by anH.225/H.235/H.245 component 132 and an SIP TSM component 134. TheH.225/H.235/H.245 component describes the H. 232 protocols suite, whereH.225 covers narrow-band visual telephone services, H.235 concernssecurity and authentication, and H.245 negotiates channel usage andcapabilities.

This protocol architecture provides a TCP/UDP layer 136, an IP4 and IP6layer 138 and 140 and at the bottom a media processing layer 142.

FIG. 9 shows the policy structure including the SIP registrar 130 andH.323 gatekeeper 122 which accesses a policy 144 including a lookupserver 146, calling plans and virtual private networks (VPN) 148, andrealms/CAC (call admission control) 150.

The architecture of FIG. 8 provides flexible mapping of the applicationas well as protocol information such as user identify (the user name andphone numbers); network topology (the host names domains) and the SIPprotocol headers (including “from”, “to”, “privacy”).

The architecture also provides full control of SIP messages, timers,state machines and call flows. This enables the issuing of messagesindependently of call participants and enables third party call control(3 pcc) which is used by various applications.

The architecture of FIG. 8 also provides full control of H.225 and H.245state machines. An inter-working with early H.245, H.245 tunneling,H.323 fast connect and H.323 v1 (which are slow start calls such asthose used by Cisco call manager) is provided. The inter-working withendpoints implementing the extended fast connect in accordance withH.460.6 (such as an Avaya PBX) is also provided.

The present architecture also offers flexible inter-working between SIPand H.323. Specifically, regular voice calls which use any of the abovefeatures as well as advanced services such as video can be inter-worked.Facsimile transmission such as T.38 fax inter-working is also supported.

The access network call admission control is used for call routing loadbalancing, rejecting calls that exceed the provisioned service levelassurance (SLA), or to provide the best effort service for overflowcalls. Call routing here refers to selecting the destination of thecall.

The multi-protocol session controller provides an enhanced calladmission control for signaling resources, including the call peers andthe call peer groups, as well as for bandwidth control for the callpeers and call peer groups. The signaling resources refer to the numberof call legs that are active on the multi-protocol session controller.

Bandwidth measurement is not based on call legs originating from orterminating on an endpoint (or realm or subnet or i-edge group). Forexample, a softswitch (an endpoint) may make a call which is hairpinnedthrough the MSC and media never leaves the endpoint. In this case, thereare two call legs (an originating and a terminating call leg) on theendpoint, while the bandwidth used is zero (inter bandwidth, which weare concerned with is zero, however, intra-bandwidth would be non-zero).

The MSC provides bandwidth control even if it is not in the media pathor in control of how the media flows in the underlying network. Mediacan either be routed directly by the MSC or controlled by using a thirdparty media server. In a case where network topology closely resemblesthe logical groups created for bandwidth control (a-d above), the MSCcan provide control of how bandwidth is used. For example, anadministrator may have fixed network resources to route media betweenSubnet1 and Subnet2 and they may be connected via an MPLS network.

In FIG. 10, a peer group is used to bundle subnets and provide calladmission control based on groups of subnets. Call admission control canbe enforced at the peer level or at the peer group level.

The provisioning of media routing policy is provided according to anembodiment of the invention. The multi-protocol session controllerallows media routing policy to be provisioned in each of the peer, thepeer-group, and the realm. At each level two separate kinds of mediarouting policy are specified, namely intra-X media routing and inter-Xmedia routing. Here X is one of the peer, the peer-group or the realm.The following describes these policies:

For peer policies under the intra-x media routing, the media routingpolicy for hairpinned calls is provided. Hairpinned calls are calls forwhich the originating and terminating peer are the same. For peerpolicies under the inter-x media routing, media routing policy for callsbetween this peer and the rest of the peers.

For peer group policies under the intra-X media routing, media routingpolicy is provided for calls where the originating and terminating peerare in the same peer group. For peer group policies under the inter-Xmedia routing, media routing policy is provided for calls where theoriginating and terminating peer are in the different peer groups. Forrealm policies under the intra-x media routing, media routing policy isprovided for calls where the originating and terminating peer are in thesame realm. For realm policies under the inter-x media routing, mediarouting policy is provided for calls where the originating andterminating peer are in the different realms.

For each of these, the policy definition consists of two values, thepolicy precedence and the policy on/off. The precedence is a numericalvalue (integer) and a higher value implies a higher precedence.

In FIG. 11, for a call from peer A to peer B, the relevant policy on thesource and destination peer is first determined. For example, the callmay be between different peers in the same peer group but betweendifferent realms. This means that on the source as well as destinationpeer, the call is subject to the following policies: inter-peer MR(media routing), intra-peer-group MR, or inter-realm MR. The followinghierarchy is then applied to determine the media routing policyapplicable to the source or destination peer. If the peer has the mediarouting (MR) policy specified, it is used. If the peer-group has mediarouting (MR) policy specified, it is used. If the realm has the mediarouting (MR) policy specified, it is used.

Once the media routing policy on each peer is determined, the precedence(the integer value) is used to determine which policy wins.

FIG. 11 provides examples of media routing policy configurations. For anuntrusted link (top example) 164 between peer A and peer C is on, thepeers A and C are on while peer B is off. For an untrusted network(bottom example) 166 where the link between A and B and between A and Care on, the peer A is on and peers B and C are off.

Far-end network address translation traversal is discussed hereinafter.The multi-protocol session controller 170, shown in FIG. 12, caninterface with any kind of generic network address translation/firewall171 (symmetric, full cone, restricted cone, etc) on a session initiationprotocol access network 172. The network address translation/firewallmay be connecting an enterprise/carrier 174, 176 and 178 to the publicinternet via a gateway 180 or to the private network 182 of a provider.

For all session initiation protocol request messages, the existence ofthe network address translation itself is detected by matching the viaheader to the source IP address of the message. The response to such arequest is always sent back to the source IP and port from which therequest came. The multi-protocol session controller also implements RFC3581, which can be used by the session initiation protocol user agentbehind the firewall to register its contact properly.

Session initiation protocol registrations are provided as follows. Themulti-protocol session controller maps the contacts registered by thesession initiation protocol endpoints behind the firewall to the sourceIP and port from which the registration comes. The multi-protocolsession controller implements the following mechanism to keep thesignaling pin-hole open.

If the multi-protocol session controller detects that the registrationis coming from behind a network address translator, the multi-protocolsession controller will tweak down the expiration timeout assigned tothe endpoint. The default timeout used will be two minutes (which can beconfigured by the admin). Some of the considerations for adopting thisapproach over others are: The suggested mechanism for network addresstranslators to refresh user datagram protocol (UDP) bindings is outboundtraffic (where traffic comes from behind the firewall, going to thenetwork address translator's public side). This is mostly for securityconcerns in that some hacker may attempt to keep a binding open longafter it has been closed.

This method may introduce a significant higher load on themulti-protocol session controller since it requires a large number ofmessages and the messages to be parsed and created on the multi-protocolsession controller. If the registrations are destined to an applicationserver, they may create a lot of load on the application server too. Themulti-protocol session controller will provide a filtering mechanism toturn down the frequency at which these registrations are sent to theapplication server. The multi-protocol session controller will alsomonitor these registrations to see if any of them need exceptionprocessing and need to be sent to the application server (for example,the callid (call ID) or contacts or any other registration uniformresource identifier/contact parameters have changed). The timeout on theapplication server side will be one day (which can be tweaked down bythe application server based on the application server configuration).For example, if the application server assigns a timeout of 60 minutes,the multi-protocol session controller will end up forwarding every 30thregistration given a two minute timeout on the network addresstranslator-side.

The multi-protocol session controller may also provide a mechanism todetect the network address translation timeout using the OPTIONSpackets. The OPTIONS refers to a command in a request pocket headerrelating to the method to be performed on the resource. Possible methodsinclude, Invite, Ach, eye, Cancel, Options, Register, as are known. Thismechanism is theoretically possible, but is not guaranteed to workdeterministically given the non-deterministic behavior of networkaddress translators. A per endpoint/subnet/realm timeout for networkaddress translation traversal is preferred.

Static endpoints are session initiation protocol gateways which do notregister are also supported. Static pinholes must be provisioned on thefirewall to let inbound signaling through.

Separation between signaling and media network address translationtraversal is far-end signaling and media network address translationtraversal is shown in FIG. 13. The multi-protocol session controllerseparates the far end NAT traversal from the signaling end NATtraversal, and is controlled through configuration. For example, themulti-protocol session controller can be deployed in the configurationshown in FIG. 13.

Independence of network address translation traversal and media routingpolicy is now discussed. Enabling network address translator traversalon an endpoint does not imply that the multi-protocol session controllerwill take control of media routing for all calls destined to/originatingfrom it.

See FIGS. 14, 15 and 16 for network address translator traversalscenarios. In each of these scenarios, the administrator would havenetwork address translator traversal enabled on the call peers. In thescenario where the endpoints are dynamic, their configuration fornetwork address translator traversal will be inherited from a globalconfiguration file or the templates. This configuration will enablesignaling network address translator traversal. The media routingconfiguration for the call peer groups which the call peers are part ofwill then be applied to determine how media must be routed.

The ring-back problem (183→200) will now be described. The numbers 183and 200 refer to message codes for responses. In the SIP, a message code183 refers to ringing of the phone being called and a message coded 200indicates that the user has picked up the phone and that the call set upis completed. The 200 OK message is sent to the call initiating phone.The multi-protocol session controller will optionally allow a SIP 183message to be signaled as a 200 OK message to the caller, to circumventthe ring-back problem. Calls for which this signaling is employed andwhich do not connect are reported in a normal fashion. The only changein the call detailed records will be that the last message sent to acaller indicates a 200 OK message. It is not advisable to do billing onthe caller side in this scenario. This feature can be configured on thetemplate ports or actual phone ports. This feature is not advised to beused for STUN (simple traversal of UDP through NATs) capable clients.This is a protocol that allows applications to discover the presence andtypes of network address translators.

Issues with network address translator traversal, include that any mediasent by the called party before the caller sends media may get clipped.The 183→200 conversion above alleviates this problem in cases where thecalled party sends media along with 183 and not before it. With the183→200 conversion described above, the multi-protocol sessioncontroller will hang up the calling party if any other final responseother than 200 is received on the called side. An appropriate reasonheader may be used in the BYE message. Note that 3xx responses comingafter the 183 will not be pursued.

Another solution to this is to use a media server/application server.The multi-protocol session controller can route the call to anapplication server after detecting that the call came from a devicebehind a network address translator. The application server will connectthe call to a media server and hunt on the second leg of the call.

Interaction with STUN/ICE based systems: The Far-End NAT traversalimplemented on the multi-protocol session controller is fully compatiblewith STUN/ICE based systems which may also use RFC 3581.

Quality of service and integration with the Juniper Networks ServiceDeployment System are described hereinafter.

The multi-protocol session controller 170 can be integrated with theJuniper Networks SDX system. The SDX enables service delivery tosubscribers over a variety of broadband access technologies like DSL,Cable etc. The SDX works with the Juniper Networks Edge Router (ERX) andallows activation of service on an as needed basis. The multi-protocolsession controller 170 could be owned by the wholesaler (who owns andadministers the core network) or the retailer (who may manage thesubscribers or services). The FIG. 17 illustrates the positioning ofmulti-protocol session controller in such a system.

The SDX Gateway 190 is a component of the SDX system which allowsexternal components to interact with the SDX components through a simpleobject access protocol (SOAP) interface 192 (the multi-protocol sessioncontroller uses the Content Provider web application). The SDX Gatewaycommunicates policy to the SAE 194 which uses common object policyservice (COPS) 196 to provision the ERX (a Juniper Networks ERX SeriesEdge Router) 198 and reserve resources for the media to flow through thenetwork.

A simplified session initiation protocol call flow is outlined. Mediawhich starts to flow before the multi-protocol session controllercommunicates the policy to the SDX may not get the right class oftreatment. For H.323, the provisioning is similar. The multi-protocolsession controller does not report media/quality of service statisticsin call detailed records in this scenario.

The reporting of quality of service and call statistics is carried outas follows. The multi-protocol session controller will report thefollowing quality of service and call statistics in order to enablequality reporting and diagnostics. These statistics will be reportedonly when multi-protocol session controller is controlling the mediaflows in the network.

Parameter Reporting basis Format Definition SIP Signaling Call-peerinteger Resettable. Indicates how message many retransmissions ofretransmissions requests and responses have occurred over an aggregateof calls done since the last reset Codec CDR string Indicates what codecwas used for the call Packet Loss Rate CDR Fixed point number Thefraction of RTP data with the binary point packets from the source atleft edge of the lost since the beginning of field. reception PacketDiscard Rate CDR Fixed point number The fraction of RTP data with thebinary point packets from the source at left edge of the that have beendiscarded field. since the beginning of reception, due to late or earlyarrival, under-run or overflow at the receiving jitter buffer. BurstDensity CDR Fixed point number The fraction of RTP data with the binarypoint packets within burst at left edge of the periods since the field.beginning of reception that were either lost or discarded. Gap DensityCDR Fixed point number The fraction of RTP data with the binary pointpackets within inter-burst at left edge of the gaps since the beginningfield. of reception that were either lost or discarded. Burst DurationCDR milliseconds The mean duration, expressed in milliseconds, of thegap periods that have occurred since the beginning of reception. Rfactor CDR integer in the range 0 The R factor is a voice to 100, with avalue quality metric describing of 94 corresponding the segment of thecall to “toll quality” and that is carried over this values of 50 orless RTP session. regarded as unusable MOS-LQ CDR on a scale from 1 toThe estimated mean 5, in which 5 opinion score for listening representsexcellent quality and 1 represents unacceptable. MOS-CQ CDR on a scalefrom 1 to The estimated mean 5, in which 5 opinion score for representsexcellent conversational quality and 1 represents unacceptable.

For voice and fax, or facsimile, transcoding the MSC can act as atranscoding engine for voice and fax calls. Transcoding services areprovided for calls handed off to an access network which requires adifferent level of compression than used by the ingress network.

The multi-protocol session controller may deploy external mediaresources to perform the transcoding. These resources may be deployed atthe access network or centralized in the network core.

As shown in FIG. 18, the multi-protocol session controller 170 providesthe following transcoding functions: G.729 4< >G.711, T.38 Fax< >G.711Pass Through fax, and DTMF (dual tone multi frequency) Transcoding (RFC2833 based DTMF<—>G.711 in band DTMF).

The transcoding function complies with the RTP Translator defined in RFC3550. The transcoding function is available for both session initiationprotocol and H.323 calls and is done on a per call basis.

Deployment of transcoding resources may be provided in the accessnetwork or core network. The multi-protocol session controller 170controls the transcoding resource by using the MSCP (Media ServicesControl Protocol) 200 which allows the transcoding resource to becontrolled by multiple multi-protocol session controllers. Themulti-protocol session controller always acts as the point where mediaenters or leaves the access network.

The multi-protocol session controller can use the following mediagateways for purposes of transcoding.

Support for 1 + 1 Support for Fax Media Gateway Redundancy TranscodingDensity Brooktrout's Yes Planned 700 Snowshore Media Firewall AudiocodesIP Media No Yes 1810-200 2000 Server 2810-300  6310-2000

The call flows are described. These flows also use the MSCP Gatewaywhich converts between MSCP and the proprietary TPNCP (Audiocodes).

Thus, the present invention provides improvements including: selectivemedia routing; call routing with layer two, layer three, codec (codingand decoding), and MOS (mean opinion score) qualifiers; an MFCP and MFCPgateway, inter-working function (IWF) and video; an integrated systemfor least cost call routing; and an integrated system for maximum profitcall routing. These features unique to embodiments of the presentinvention and provide significant differentiation to this technology.

The mean opinion score (MOS) is a scale that determines relative qualityof voice communications as subjectively perceived by human userslistening to speech over a communications network. One way delay andsignal loss are significant factors in the mean opinion score, althoughother factors affect the perception of the human user as well.

Selective media routing is provided. The present technology allows theseparation of the signaling and bearer networks. However, typically, thesignaling and/or bearer are forced through certain network elements tomonitor and enforce quality of service, optimize network route etc.

Selective media routing provides control to the network service providerto do media routing on a dynamic basis, with the least amount ofconfiguration. Selective media routing policies are based on eitheringress or egress call peer's policies. The criteria for both precedencevalues and on/off is network design, based on the creation of trustboundaries. If the peers are all inside the network operators trustboundary, then there is no need to do media routing. Hence, most of thepeers that are designed to handle calls within the trust boundary willhave the default value for the policy set to “OFF”. However, peers atthe edge of the network can potentially have calls routed to them fromnon-trusted networks. When calls come from non-trusted networks, thensuch calls should be media routed. So, a precedence value is set in theedge peers; if the value of the precedence value is lower than thenon-trusted peer's precedence value, then the media routing policy ofthe non-trusted peer takes over and media routing happens.

See FIG. 20 for an example wherein the multi-protocol session controller170 is connected at a trust boundary 240 to provide media communicationsacross the trust boundary to an endpoint C 242. Endpoints A and B 244and 246 are within the trust boundary 240. The precedence for endpoint Cis at 10 with the value set to ON, while the precedence for endpoint A244 is at 5 with the value OFF, and for endpoint B is at 5 with thevalue OFF.

The following table provides an overview of whether media routing isused in communications between trusted and non-trusted networks.

Non-trusted network Trusted network Always media route Non-trustednetwork Non-trusted network Always media route Trusted network Trustednetwork Do not media route Trusted network Non-trusted network Alwaysmedia route

Turning to FIG. 19, the figure shows the physical architecture of themulti-protocol session controller 170. A dual CPU processing unit 210 isliked with a network processor cart 212. Operating components include acall processing unit 214 with a firewall control entity (FCE) 216 as aninterface to a media firewall control protocol (MFCP) 218. The MFCP 218communicates through an MFCP server 220 to a session filter on iXP2400,222, that enables separation of signaling and bearer channels.

The Media Routing functionality has been described elsewhere in thisspecification.

Call routing with qualifiers will now be described. The traditionalconcept of call routing involves making decisions on which trunk to“switch” on based on the dialed number for the call, and/or eoriginating trunk. However, in VoIP based call routing, the presentinvention has extended that notion to include call routing with layertwo identifiers such as VLAN IDs (Virtual LAN Identifiers), with layerthree identifiers such as Diffserv/TOS markings, with codec (coding anddecoding device) preferences for the call, and with mean opinion scores(MOS) qualifier for previous calls to and/or from thatdestination/origination, etc. The present call routing can ensure a highquality of service of the call by controlling the call routing.

In call routing with qualifiers, criteria are used for identifying layer2 and layer 3 components. The layer 2 and layer 3 qualifiers are used toidentify the source as well as set a marker for the egress network touse for its quality of service. Typical layer 2 qualifiers are VLAN tagsand priority bits. The VLAN tags are specified in IEEE Standard 802.1Qand the priority bits are specified in IEEE Standard 802.1p. Using theVLAN tags, the present multi-protocol session controller can identifythe source of the call and media as from a given call peer. Once theingress call peer is identified, then appropriate policies can beapplied to the call. Layer 3 identifiers are typically a uniquelyidentifiable IP address and IP subnet addresses.

The mean opinion score (MOS) is used by the multi-protocol sessioncontroller and is derived by looking at the incoming media stream andmaking measurements of jitter, latency and packet loss. The computationof the MOS score is based on the ITU-T standard E-model (G.107). FIG. 21illustrates the voice quality measurement for MOS and E-model. Themulti-protocol session controller 170 is provided between endpoint A 250on an access network 252 and an endpoint B 254 on a provider network256. Measurements of call quality are based on jitter and packet lossand are made as forward measurements for communications from endpoint A250 to endpoint B 254 and as reverse measurements for communicationsfrom endpoint B 254 to endpoint A 250.

A media firewall control protocol (MFCP) gateway is provided. Toseparate signaling and media networks and to scale signaling and mediaindependently, a “control” protocol was specified. This control protocolis referred to as Media Firewall Control Protocol (MFCP) 218. The mediafirewall control protocol can be used to control firewalls, mediaservers and also edge routers. However, firewalls, media servers andedge routers may have implemented their own control protocol forupdating policies on their system. The present invention provides alogical entity called the MFCP Gateway that takes MFCP as an input andconverts it to the appropriate control protocol of the firewall, mediaserver or edge router.

An interworking function (IWF) and video is provided. The interworkingFunction (IWF) involves the conversion between SIP and H.323 call setupprotocols used widely for setting up calls in the VoIP arena. Mappingbetween SIP and H.323 is not very straightforward, and has heretoforebeen loosely specified. The present technology provides for seamlessconversion between SIP and H.323, and vice versa. However, video callsor calls that involves sending video and audio as media is a hardproblem to solve. The mapping of video capabilities between SIP andH.323 is not well understood and not well documented. The presenttechnology encompasses SIP-H.323 IWF for video calls also.

The present invention provides an integrated system or least cost callrouting. The cost of call routing includes a number of factors,including the actual dollar cost of buying and selling routes, as wellas the quality parameters such as post dial delay (PDD), answer seizureratio (ASR), mean opinion score (MOS), and others. The dollar cost ofbuying and selling routes is known by the network administrator and isinput by the network administrator for utilization by the sessioncontroller. The quality factors may be measured and the measurementsutilized by the session controller. The present session controllerutilizes this hybrid notion of cost which includes the networkoperator's actual cost of doing business (on that route) as well as theuser experience (as measured by the quality parameters such as PDD, ASR,MOS, etc), so that the network operator can have a very optimizednetwork for both profit and operations.

Network operators, whose primary service to their customers is networktransit, find that their cost for carrying a telephone call depends uponwhere the VoIP call is destined and is variable depending upon the paththat that call takes through their network as well as through thenetworks of their partners. Hence, network operators seek to transitevery call through their network using the technique called least costrouting (LCR).

The present session controller along with the iVMS provides a method fordoing LCR. The present session controller routes all VoIP calls based onpolicies setup by the network operators. The iVMS system actually takesin rates for various paths through the network, and can compute the bestset of policies that result in least cost routing of every call that thepresent session controller processes.

The present invention provides an integrated system for maximum profitcall routing. Network operators, once they have the right policies fordoing least cost routing (LCR), also have to consider the profit thatthey make on every call that passes through their network. To maximizeprofit, they have to consider not only least cost routing for carriageor termination through their network, but also the origination income.This mode of operation involves looking at aggregate cost oftermination, and provides a policy layer on top of LCR, but which canalso be sometimes orthogonal to it.

The present session controller along with iVMS provides a method fordoing maximum profit routing (MPR). The present session controllerroutes all VoIP calls based on policies setup by the network operators.The iVMS system takes as input, dollar rates not only for varioustransit and/or terminations, but also rates for originations and thencan compute the appropriate policies that result in MPR. These policiesare then input into the session controller directly from the iVMS,resulting in MPR for the network operator.

In another feature of a preferred embodiment, the system uses anin-memory database. In FIG. 22, the multi-protocol session controllerpolicy database (referred to hereinafter as simply the “database”) isstored in two forms on a runtime multi-protocol session controller, as apersistent database on the hard drive disk and in-memory in RandomAccess Memory (RAM) of the computing platform used to run themulti-protocol session controller. When the multi-protocol sessioncontroller application is started, the persistent database (“P-DB”) onthe disk is read and stored in memory (“M-DB”) for very fast querying ofthe policy information required to process each call handled by themulti-protocol session controller.

Every call handled by the multi-protocol session controller requirespolicy lookups. The in-memory database, M-DB, provides a repository forpolicy information, required to process calls. Since very highperformance is required of the multi-protocol session controller, thesepolicy lookups must possess the following properties: The query timemust be minimal, and must not change under the same load conditions dueto other activity in the system, and the query time should not increaselinearly or exponentially with the increase in the number ofsimultaneous calls being handled.

The in memory database M-DB was designed to satisfy these constraints.The in memory database is organized into multiple “database tables” tostructure the policy data. The in memory database does not expose astandard query language interface such as SQL, to other programs. Aprogrammatic application programming interface (API) provides atomicoperations on the persistent database P-DB, which is used by thecomponents of the multi-protocol session controller, such as the GIS(call processor), the Jserver (provisioning agent) and the CLI (commandline interface). In one embodiment, the in memory database includes thefollowing tables call routes table, endpoint table, call plan bindingtable, call admission control (CAC) table, triggers table, VPN table,and realms table.

In order to query the database as quickly as possible, each table isindexed multiple times, resulting in multiple “keys” (in traditionaldatabase parlance). The innovative aspect of this table structure isthat each key does not have to be unique, even though some keys areunique, such as phone numbers, for example. The search methodology foreach of these keys could be different. The search methodologies usedwithin the in memory database M-DB include: binary search, hash search,and ternary search tree.

In particular, see the following:

Search Algorithm Algorithm Efficiency Binary search 0 (log n) Hashsearch 0 (1) Ternary search tree 0 (constant * log n)

According to an aspect of the present system, asynchronous write-throughis provided. The persistent database P-DB provides persistence to theinformation stored in the in memory database M-DB. Hence, theinformation in the M-DB and the P-DB should be identical and cannot getout of synchronization for too long. At any point in time, in anoperational multi-protocol session controller, the in memory databaseM-DB will hold the more authoritative information than the persistentdatabase P-DB. As such, whatever information is written into thepersistent database P-DB must also be written into the in memorydatabase M-DB and vice versa. However, there are several processes thatread from and update the persistent database P-DB. A common ApplicationProgramming Interface (API) is used to interface to the database. TheAPI to the persistent database P-DB is used by the CLI (command lineinterface), GIS (call processor) and the Jserver (provisioning agent)processes.

This is a process whereby the persistent database P-DB is updated via a“write-through” of the in memory M-DB. The in memory database M-DB isalways updated first before persistent database P-DB is-updated.

The API to the persistent database P-DB updates the in memory databaseM-DB database transparently. The API also updates the in memory databaseM-DB first, before updating the persistent database P-DB. The sideeffect of this is that multiple commits to the same table of the M-DBcan happen before the P-DB is actually updated. In such cases, thepersistent database P-DB will only have the last and most authoritativeupdate committed. This is achieved by the API using two principles:asynchronicity and data independent update

In updating using asynchronicity, the API performs an asynchronousupdate of the information. Any information given to the API forcommitting into the database is first updated into the in memorydatabase M-DB and then queued for update to the persistent databaseP-DB. The queuing is necessary as the disk update requires operatingsystem scheduling intervention, and a bulk update of the disk is moreefficient than multiple sporadic writes to the disk. However, theinformation is already committed to the in memory database M-DB andhence is available to all the processes for querying.

The API when updating the in memory database M-DB, only uses the keys tothe table and is not aware of the data itself. In traditional databaseStructured Query Language (SQL) based systems, the commit commandcarries the data to be updated too. However, the API here is only awareof the keys and not the data itself. The data is opaque to the API. Assoon as the API uses the key to find the correct entry, the commit ofthe data happens in one operation. If a subsequent commit operates onthe same key, then the data in the in memory database M-DB will getupdated again, even before the commit queue to the persistent databaseP-DB is completed.

The asynchronous write-through procedure of the database provides anumber of benefits, including that the information is always availablefor high performance applications, the integrity of information andsequentiality is maintained, and the persistence of information istransparently maintained.

Thus, there is provided a system and method for voice and real time orat least nearly real time communications over a packet switched network.The present system includes a multi-protocol session controller that canbe deployed as either a core controller or a border controller. Thepresent session controller provides selective media routing; callrouting with layer two, layer three, codec (coding and decoding), andMOS (mean opinion score) qualifiers; an MFCP and MFCP gateway;inter-working function (IWF) and video; an integrated system for leastcost call routing; an integrated system for maximum profit call routing;and an in memory database.

Although other modifications and changes may be suggested by thoseskilled in the art, it is the intention of the inventors to embodywithin the patent warranted hereon all changes and modifications asreasonably and properly come within the scope of their contribution tothe art.

We claim:
 1. An apparatus, comprising: a session controller configuredto be coupled to a communications network and to provide services toendpoints in the communications network, the session controllerconfigured to receive an indication of a Voice over Internet Protocol(VoIP) registration request at a first time from an endpoint in thecommunications network, the endpoint to be registered at an applicationserver, the session controller being further configured to receive asignaling message from the endpoint at a second time later than thefirst time, the signaling message for requesting a service, the sessioncontroller being further configured to forward an indication of the VoIPregistration request and the signaling message to the applicationserver, the session controller being further configured to forward anindication of the VoIP registration request and the signaling messagesuch that at least a portion of the topology of the communicationsnetwork beyond the session controller is hidden to the endpoint, whereinthe at least a portion of the topology hidden includes an identity ofthe application server; wherein the session controller is configured toforward the indication of the VoIP registration request and thesignaling message to the application server based on a uniform resourceidentifier; wherein the session controller is configured to forward theindication of the VoIP registration request and the signaling messagetransparently such that the identity of the application server isunidentifiable by the endpoint; and wherein the session controllercomprises at least one processor.
 2. The apparatus of claim 1, whereinthe session controller is configured to define a dynamic endpoint stateon a realm of the application server and to apply a media routingcharacteristic of the realm to a call originating from or terminating atthe endpoint.
 3. The apparatus of claim 1, wherein the sessioncontroller is configured to process a SIP 3XX redirect message code. 4.A computer program stored on a non-transitory computer-readable medium,the computer program comprising: a first receiving instruction toreceive, at a session controller coupled to a communications network forproviding services to endpoints in the communications network, anindication of a VoIP registration request from an endpoint in thecommunications network, the endpoint to be registered at a third partyapplication server; a second receiving instruction to receive, at thesession controller, a signaling message from the endpoint at a secondtime later than the first time, the signaling message for requesting aservice; a forwarding instruction to transparently forward an indicationof the VoIP registration request and the signaling message to anapplication server such that at least a portion of the topology of thecommunications network beyond the session controller is hidden to theendpoint, wherein the at least a portion of the topology hidden includesan identity of the application server; wherein the forwardinginstruction is further to forward transparently such that the identityof the application server is unidentifiable by the endpoint; and whereinthe forwarding instruction is further to forward based on a uniformresource identifier of the application server.
 5. The computer programstored on a computer-readable medium of claim 4, the computer programfurther comprising a definition instruction to define a dynamic endpointstate on a realm of the application server and further to apply a mediarouting characteristic of the realm to a call originating from orterminating at the endpoint.
 6. The computer program stored on acomputer-readable medium of claim 4, the computer program furthercomprising a processing instruction to process a SIP 3xx redirectmessage code.